apparatus for processing a signal and method thereof

ABSTRACT

An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving coding mode information indicating a speech coding scheme or an audio coding scheme, linear prediction coding degree information indicating a linear prediction coding degree, and the signal including at least one of a speech signal and an audio signal; decoding the signal according to the speech coding scheme or the audio coding scheme based on the coding mode information; decoding linear prediction coding coefficients of the signal based on the linear prediction coding degree information; and generating an output signal by applying the decoded linear prediction coding coefficients to the decoded signal. In this case, the linear prediction coding degree information is determined based on a variation of a value of an LPC residual generated from performing the linear prediction coding on the signal.

TECHNICAL FIELD

The present invention relates to an apparatus for processing a signaland method thereof. Although the present invention is suitable for awide scope of applications, it is particularly suitable for simplifyinga structure of a coding device by performing LPC (linear predictioncoding) using a variable degree.

BACKGROUND ART

Generally, an audio property based coding scheme is used for such anaudio signal as a music signal. A speech property based coding scheme isused for a speech signal. If an audio signal is included in a speechsignal more or less, it is able to use a coding scheme of a frequencydomain processing based on the speech property.

DISCLOSURE OF THE INVENTION Technical Problem

However, a speech and audio signal processor according to a related artuses three kinds of modules for performing three kinds of the abovementioned coding schemes. As the number of the usable modules isincremented, if an inter-module switching is generated, the number oftransition parts to be processed is incremented as well.

Technical Solution

Accordingly, the present invention is directed to an apparatus forprocessing a signal and method thereof that substantially obviate one ormore of the problems due to limitations and disadvantages of the relatedart.

An object of the present invention is to provide an apparatus forprocessing a signal and method thereof, by which distortion of anoriginal signal is prevented using linear prediction coding of avariable degree instead of a fixed degree.

Additional features and advantages of the invention will be set forth inthe description which follows, and in part will be apparent from thedescription, or may be learned by practice of the invention. Theobjectives and other advantages of the invention will be realized andattained by the structure particularly pointed out in the writtendescription and claims thereof as well as the appended drawings.

To achieve these and other advantages and in accordance with the purposeof the present invention, as embodied and broadly described, a method ofprocessing a signal according to the present invention includesreceiving the signal including at least one of a speech signal and anaudio signal, coding mode information indicating a speech coding schemeor an audio coding scheme and linear prediction coding degreeinformation indicating a linear prediction coding degree, decoding thesignal according to the speech coding scheme or the audio coding schemebased on the coding mode information, decoding linear prediction codingcoefficients of the signal based on the linear prediction coding degreeinformation, and generating an output signal by applying the decodedlinear prediction coding coefficients to the decoded signal. And, thelinear prediction coding degree information is determined based on avariation of a value of an LPC residual generated from performing thelinear prediction coding on the signal.

Preferably, if the signal is the speech signal having a lot of voicedsound, the linear prediction coding degree information indicates adegree higher than that of the speech signal having a lot of unvoicedsound.

Preferably, if the signal is the audio signal having a strong tonalcomponent, the linear prediction coding degree information indicates adegree higher than that of the audio signal having a week tonalcomponent.

Preferably, if the signal is an audio-like signal, the linear predictioncoding degree information indicates a degree lower than that of aspeech-like signal.

Preferably, a frame length of the audio signal is an integer multiple ofa frame length of the speech signal.

Preferably, the method further includes, if the decoded signal is thespeech or audio signal and a signal of a previous frame is differentfrom the decoded signal, preventing aliasing by compensating the decodedsignal using the signal of the previous frame.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing a signalincludes a multiplexer receiving an LPC bitstream including linearprediction coding degree information indicating a linear predictioncoding degree and linear prediction coding coefficients, the signalincluding at least one of a speech signal and an audio signal, andcoding mode information indicating a speech coding scheme or an audiocoding scheme, an ACELP decoding unit decoding the signal according tothe speech coding scheme if the coding mode information indicates thespeech coding scheme, a TCX decoding unit decoding the signal accordingto the audio coding scheme if the coding mode information indicates theaudio coding scheme, and an LPC decoding unit decoding the linearprediction coding coefficients of the signal based on the linearprediction coding degree information, the LPC decoding unit generatingan output signal by applying the decoded linear prediction codingcoefficients to the decoded signal. Moreover, the linear predictioncoding degree information is determined based on a variation of a valueof an LPC residual generated from performing the linear predictioncoding on the signal.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, a method of processing a signalincludes determining linear prediction coding degree informationindicating a variable degree of linear prediction coding coefficientsaccording to property of an input signal, determining the linearprediction coding coefficients based on the linear prediction codingdegree information, generating an LPC residual using the linearprediction coding coefficients and the input signal, and coding the LPCresidual using either an audio coding scheme or a speech coding scheme.Moreover, the linear prediction coding degree information is determinedbased on a variation of a value of an LPC residual generated fromperforming the linear prediction coding on the input signal.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing a signalincludes an LPC analysis unit determining linear prediction codingdegree information indicating a variable degree of linear predictioncoding coefficients according to property of an input signal, the LPCanalysis unit determining the linear prediction coding coefficientsbased on the linear prediction coding degree information, the LPCanalysis unit generating an LPC residual using the linear predictioncoding coefficients and the input signal, an ACELP encoding unit codingthe LPC residual using an audio coding scheme, a TCX encoding unitcoding the LPC residual using a speech coding scheme, and a multiplexergenerating a bitstream including the linear prediction coding degreeinformation, the linear prediction coding coefficients and the codedsignal. Moreover, the linear prediction coding degree information isdetermined based on a variation of a value of an LPC residual generatedfrom performing the linear prediction coding on the input signal.

It is to be understood that both the foregoing general description andthe following detailed description are exemplary and explanatory and areintended to provide further explanation of the invention as claimed.

Advantageous Effects

Accordingly, the present invention provides the following effects oradvantages.

First of all, the present invention variably determines an applicationdegree of linear prediction coding using linear prediction degreeinformation variably determined according to a property of a signal,thereby efficiently coding a signal resulting from mixing speech andaudio signals together using modules of which number is smaller thanthat of modules configuring a general signal processor.

Secondly, the present invention performs linear prediction coding byvariably determining linear prediction degree information according to aproperty of a signal, thereby preventing signal distortion frequentlygenerated in case of performing linear prediction coding of a fixeddegree. And, the present invention is able to efficiently code a speechor audio signal of which LPC modeling is difficult.

DESCRIPTION OF DRAWINGS

The accompanying drawings, which are included to provide a furtherunderstanding of the invention and are incorporated in and constitute apart of this specification, illustrate embodiments of the invention andtogether with the description serve to explain the principles of theinvention.

In the drawings:

FIG. 1 is a schematic block diagram of an apparatus for processing asignal including speech and audio signals according to a firstembodiment of the present invention;

FIG. 2 is a schematic block diagram of a signal encoding apparatusaccording to a second embodiment of the present invention;

FIG. 3 is a detailed block diagram of an LPC analysis unit of a signalencoding apparatus according to a second embodiment of the presentinvention;

FIG. 4 is a schematic block diagram of a signal decoding apparatuscorresponding to the signal encoding apparatus shown in FIG. 2;

FIG. 5 is a detailed block diagram of an LPC synthesis unit of a signaldecoding apparatus according to a second embodiment of the presentinvention;

FIG. 6 is a flowchart of a signal decoded in accordance with linearprediction degree information in an LPC synthesis unit according to asecond embodiment of the present invention;

FIG. 7 is a schematic block diagram of a signal decoding apparatus iflinear prediction degree information of the present invention indicates0;

FIG. 8 is a diagram for various examples of applying a different codingscheme per frame (or subframe) to an input signal configured by a blockunit according to the present invention;

FIG. 9 is a diagram of frames (or subframes), to which different codingschemes are applied, among contiguous frames (or subframes);

FIG. 10 is a diagram for a processing method in case that windows ofdifferent types are overlapped in the frames, to which the differentcoding schemes are applied, respectively;

FIG. 11 is a schematic block diagram of a signal encoding apparatusaccording to another embodiment of the present invention;

FIG. 12 is a schematic block diagram of a signal decoding apparatusaccording to another embodiment of the present invention;

FIG. 13 is a schematic block diagram of a product in which a signaldecoding apparatus according to the present invention is implemented;and

FIG. 14 is a diagram for explaining relations between products in whicha signal processing apparatus according to the present invention isimplemented.

BEST MODE

Additional features and advantages of the invention will be set forth inthe description which follows, and in part will be apparent from thedescription, or may be learned by practice of the invention. Theobjectives and other advantages of the invention will be realized andattained by the structure particularly pointed out in the writtendescription and claims thereof as well as the appended drawings.

To achieve these and other advantages and in accordance with the purposeof the present invention, as embodied and broadly described, a method ofprocessing an audio includes receiving coding mode informationindicating a speech coding scheme or an audio coding scheme, linearprediction coding degree information indicating a linear predictioncoding degree, and the signal including at least one of a speech signaland an audio signal; decoding the signal according to the speech codingscheme or the audio coding scheme based on the coding mode information;decoding linear prediction coding coefficients of the signal based onthe linear prediction coding degree information; and generating anoutput signal by applying the decoded linear prediction codingcoefficients to the decoded signal, wherein the linear prediction codingdegree information is determined based on a variation of a value of anLPC residual generated from performing the linear prediction coding onthe signal.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing a signalincludes a multiplexer receiving an LPC bitstream including linearprediction coding degree information indicating a linear predictioncoding degree and linear prediction coding coefficients, the signalincluding at least one of a speech signal and an audio signal, andcoding mode information indicating a speech coding scheme or an audiocoding scheme; an ACELP decoding unit decoding the signal according tothe speech coding scheme if the coding mode information indicates thespeech coding scheme; a TCX decoding unit decoding the signal accordingto the audio coding scheme if the coding mode information indicates theaudio coding scheme; and an LPC synthesis unit decoding the linearprediction coding coefficients of the signal based on the linearprediction coding degree information, and generating an output signal byapplying the decoded linear prediction coding coefficients to thedecoded signal, wherein the linear prediction coding degree informationis determined based on a variation of a value of an LPC residualgenerated from performing the linear prediction coding on the signal.

It is to be understood that both the foregoing general description andthe following detailed description are exemplary and explanatory and areintended to provide further explanation of the invention as claimed.

Mode for Invention

Reference will now be made in detail to the preferred embodiments of thepresent invention, examples of which are illustrated in the accompanyingdrawings. First of all, terminologies or words used in thisspecification and claims are not construed as limited to the general ordictionary meanings and should be construed as the meanings and conceptsmatching the technical idea of the present invention based on theprinciple that an inventor is able to appropriately define the conceptsof the terminologies to describe the inventor's invention in best way.The embodiment disclosed in this disclosure and configurations shown inthe accompanying drawings are just one preferred embodiment and do notrepresent all technical idea of the present invention. Therefore, it isunderstood that the present invention covers the modifications andvariations of this invention provided they come within the scope of theappended claims and their equivalents at the timing point of filing thisapplication.

Specifically, ‘coding’ can be construed as ‘encoding’ or ‘decoding’selectively.

Moreover, ‘information’ in this disclosure is the terminology thatgenerally includes values, parameters, coefficients, elements and thelike and its meaning can be construed as different occasionally, bywhich the present invention is non-limited.

In this disclosure, in a broad sense, an audio signal is conceptionallydiscriminated from a video signal and designates all kinds of signalsthat can be auditorily identified in case of being played back. In anarrow sense, the audio signal means a signal having none or smallquantity of speech properties. Audio signal of the present inventionshould be construed in a broad sense. Yet, the audio signal of thepresent invention can be understood as an audio signal in a narrow sensein case of being used as discriminated from a speech signal.

FIG. 1 is a schematic block diagram of an apparatus 100 and 200 forprocessing a signal including speech and audio signals according to afirst embodiment of the present invention.

Referring to FIG. 1, a signal processing apparatus according to a firstembodiment of the present invention can include an encoder 100 and adecoder 200. And, the encoder 100 can mainly include a signal classifier110, a first encoding unit 120-1, a second encoding unit 120-2 and amultiplexer 170.

The signal classifier 110 analyzes properties of an input signal,determines what kind of coding scheme is used to encode a current frame(or subframe) based on the analyzed properties, and is then able togenerate coding mode information indicating the determined codingscheme. In this case, the generated coding mode information indicates afirst coding scheme used by the first encoding unit 120-1 and a secondcoding scheme used by the second encoding unit 120-2, or is able toindicate one of an A coding scheme used by an A encoding unit 130, a Bcoding scheme used by a B encoding unit 150 and a C coding scheme usedby a C encoding unit 160. The coding schemes shall be described indetail later in this disclosure, by which the present invention isnon-limited.

The first encoding unit 120-1 and the second encoding unit 120-2indicate units, to which the first and second coding schemes based ondifferent domains are applied, respectively. In this case, the domainscan include a linear prediction coding (LPC) domain, a frequency domain,a time domain and the like. For instance, if the first coding schemeindicates the coding scheme based on the linear prediction codingdomain, the second coding scheme is able to indicate the coding schemebased on the frequency domain. Regarding this, definitions andproperties according to domain types shall be described in detail later.

In case that the second coding scheme indicates the linear predictioncoding domain based coding scheme, the second encoding unit 120-2 caninclude a linear prediction coding (LPC) analysis unit 140 and a pair ofspecific coding units 150 and 160 to which different coding schemes areapplied, respectively.

The linear prediction coding analysis unit 140 performs linearprediction coding on an inputted signal to generate linear predictioncoding coefficients and an LPC residual remaining after the prediction.In this case, it is preferable that the linear prediction codingcoefficient is generally fixed to 16 degrees in general.

Afterwards, the LPC residual is inputted to the B encoding unit 150 orthe C encoding unit 160 and is then encoded by the B or C coding scheme.In this case, the B coding scheme can include ACELP (algebraic codeexcited linear prediction) and the C coding scheme can include TCX(transform coded excitation), by which the present invention isnon-limited. Meanwhile, in case that the B encoding unit 150 and the Cencoding unit 160 use ACELP and TCX, respectively, the A encoding unit130 preferably uses MDCT (modified discrete Fourier transform).Generally, ACELP is the coding scheme suitable for a speech signal,while MDCT or TCX is the coding scheme suitable for an audio signal. Ageneral signal processor uses MDCT to process a signal of which majorcomponent is an audio signal and uses TCX to process a small quantity ofan audio signal included in a signal consisting of the audio signal anda speech signal corresponding to a major component of the signal.

Referring to FIG. 1, the coding mode information can include codingidentity information and sub-coding identity information. The codingidentity information indicates either the first coding scheme or thesecond coding scheme applied to a current frame. And, in case that acurrent frame uses the second coding scheme, the sub-coding identityinformation may be the information indicating whether the B codingscheme or the C coding scheme is used.

The coding mode information is generated by the signal classifier 110,determines the coding unit 120-1/120-2 or 130/150/160 to which the inputsignal will be inputted, and is then transmitted to the multiplexer 170.

Meanwhile, as mentioned in the foregoing description, the input signalis partitioned per frame or subframe based on the coding modeinformation generated by the signal classifier 110 and is then inputtedto the first encoding unit 120-1 or the second encoding unit 120-2.Subsequently, data encoded by each coding scheme according to the abovedescribed method can be transmitted to the multiplexer 170.

The multiplexer 170 generates at least one or more bitstreams bymultiplexing the coding mode information and the data encoded by each ofthe coding units together and then transmits the generated at least oneor more bitstreams.

The decoder 200 of the signal processing apparatus according to thefirst embodiment of the present invention is able to mainly include ademultiplexer 210, a first decoding unit 220-1 and a second decodingunit 220-2. In this case, the first decoding unit 220-1 and the seconddecoding unit 220-2 are the components of a decoder side to correspondto the first encoding unit 120-1 and the second encoding unit 120-2described with reference to the encoder 100, respectively. As mentionedin the foregoing description with reference to the encoder 100, thedecoder 200 can include an A decoding unit 230, a B decoding unit 240and a C decoding unit 250. If the first decoding unit 220-1 correspondsto the A decoding unit 230, the second decoding unit 220-2 can includethe B decoding unit 240 and the C decoding unit 250. Moreover, thesecond decoding unit 220-2 is able to further include an LPC synthesisunit 270 configured to reconstruct an original signal using the linearprediction coding coefficients by receiving an input of a signal decodedby the B decoding unit 240 or the C decoding unit 250.

A first decoding scheme applied by the first decoding unit 220-1 and asecond decoding scheme applied by the second decoding unit 220-2indicate the decoder side schemes corresponding to the coding schemesdescribed with reference to the encoder 100. Moreover, in case that thedecoder 200 includes the A decoding unit 230, the B decoding unit 240and the C decoding unit 250, A to C decoding schemes respectivelyapplied to the A to C decoding units 230 to 250 indicate the decodingschemes corresponding to the coding schemes described with reference tothe encoder 100 as well. In particular, the A decoding scheme is MDCT,the B decoding scheme is ACELP, and the C decoding scheme is TCX, forexample. And, the decoding schemes can be determined according to thecoding mode information extracted by the demultiplexer 210. As mentionedin the foregoing description, the coding mode information can includecoding identity information and sub-coding identity information as well.

Thus, the signal processing apparatus according to the first embodimentof the present invention mainly uses two kinds of modules (i.e., thefirst coding unit 220-1 and the second coding unit 220-2) according todomains processed in accordance with signal properties. In particular,the signal processing apparatus according to the first embodiment of thepresent invention includes the module (i.e., the B decoding unit 240)configured to process a speech signal if an input signal includes thespeech signal. And, the signal processing apparatus according to thefirst embodiment of the present invention includes the module (i.e., theC decoding unit 250) configured to process an audio signal included inthe speech signal. Moreover, the signal processing apparatus accordingto the first embodiment of the present invention includes the module(i.e., the A decoding unit 230) configured to process an audio signal ifan input signal includes the audio signal.

So to speak, the signal processing apparatus according to the firstembodiment of the present invention includes three kinds of modulesincluding an MDCT module similar to a conventional AAC to process anaudio signal, an ACELP module configured to process a speech signal inLPD, and a TCX module configured to process an audio signal in LPD.Since each module uses a different window type, a different frame lengthand the like, if a module switching occurs, a method of processingtransition is further required.

Therefore, a second embodiment of the present invention proposes asignal process apparatus capable of reducing signal distortion in codingby decrementing the number of modules used for signal processing.

FIG. 2 is a schematic block diagram of a signal encoding apparatus 100Aaccording to a second embodiment of the present invention.

Referring to FIG. 2, a signal encoding apparatus 100A mainly includes anLPC analysis unit 240, an ACELP encoding unit 250, a TCX encoding unit260 and a multiplexer 270. And, the signal encoding apparatus 100A isable to further include a signal classifier (not shown in the drawing).In this case, the signal classifier determines a coding scheme based onproperty of an input signal, generates coding mode informationindicating the coding scheme, and then transmits the generated codingmode information to the multiplexer 270.

The LPC analysis unit 240 receives an input signal and then performslinear prediction coding on the received input signal. In doing so, theLPC analysis unit 240 according to the second embodiment of the presentinvention is able to generate a coefficient of a variable degreeaccording to property of the input signal, whereas the former LPCanalysis unit 140 of the encoder 100 described with reference to FIG. 1performs the linear prediction coding to generate a linear predictioncoding coefficient of a fixed degree, e.g., 16 degree. The LPC analysisunit 240 calculates an LPC residual by performing the linear predictioncoding of various degrees on the input signal and is then able todetermine the variable degree based on the variation of the LPC residualsignal.

In other words, as the linear prediction coding degree increases toreduce the LPC residual, it means that the corresponding input signal isfit for the linear prediction of high degree. Hence, the high degree canbe determined as the variable degree for the linear prediction coding.On the contrary, if the LPC residual is not reduced despite performingthe linear prediction coding by raising the linear prediction codingdegree, it is determined that the linear prediction coding of the highdegree is not preferable for the input signal, it is able to determine alow degree as the variable degree for the linear prediction coding.Based on whether modeling the input signal is performed well using thelinear prediction coding (i.e., whether the LPC residual is reduced),linear prediction codings of various degrees are performed anddetermined. Hence, it is able to determine the variable degreecorrespondingly.

In general, in case of a speech signal, since LPC modeling of a signalhaving a lot of voiced sound is performed better than LPC modeling of asignal having a lot of unvoiced sound, it is able to use the linearprediction coding degree of higher degree. In case of an audio signal, atonal-strong signal is able to use a linear prediction coding degreehigher than a degree of a noise-like signal.

For another instance, if an input signal is a signal in which propertyof a speech signal is dominant (hereinafter, such a signal is named aspeech-like signal), a linear prediction coding coefficient of higherdegree is generated. If an input signal is a signal in which property ofan audio signal is dominant (hereinafter, such a signal is named anaudio-like signal), a linear prediction coding coefficient of low degreecan be generated.

Besides, the LPC analysis unit 240 configured to perform the linearprediction coding for generating a coefficient of variable degree shallbe described in detail with reference to FIG. 3 later in thisdisclosure.

The LPC analysis unit 240 is able to generate linear prediction coding(LPC) degree information, linear prediction coding (LPC) coefficientsand LPC residual by performing the linear prediction coding on the inputsignal. In doing so, as mentioned in the foregoing description, the LPCdegree information may be variable according to property of the inputsignal.

The linear prediction coding degree information and the linearprediction coding coefficients generated by the LPC analysis unit 240are transmitted to the multiplexer 270. And, the LPC residual isinputted to the ACELP encoding unit 250 or the TCX encoding unit 260based on the coding mode information (not shown in he drawing)determined by the signal classifier (not shown in the drawing). In thiscase, unlike the former coding mode information described with referenceto FIG. 1, the coding mode information can indicate either the ACELPcoding scheme or the TCX coding scheme as shown in Table 1.

TABLE 1 coding mode information (coding_mode) meaning 0 Coding by ACELPcoding scheme 1 Coding by TCX coding scheme

If the coding mode information indicates that the ACELP coding scheme isused, the ACELP encoding unit 250 is able to encode a signal by thescheme determined by receiving an input of the LPC residual. The ACELPencoding unit 250 has the same function of the B encoding unit 150described with reference to FIG. 1, of which details are omitted fromthe following description. Moreover, in this disclosure, the ACELPcoding scheme corresponds to the voice coding scheme.

The TCX encoding unit 260 mainly includes an MDCT unit 261, a quantizer262 and an entropy encoding unit 263. In this disclosure, the TCX codingscheme can be called an audio coding scheme. The MDCT unit 261 receivesan input of the LPC residual and then performs MDCT on a signal. Thetransformed signal is inputted to the quantizer 262 and the quantizationis performed. As used in AAC, frequency bands are grouped into scalefactor bands (sfb) that use the same scale factor. The scale factorbands are then used. Moreover, the quantizer 262 receives an input of amasking threshold per frequency band calculated from an original inputsignal prior to being inputted to the LPC analysis unit 240 and is thenable to perform the quantization per the scale factor band withreference to the masking threshold. In doing so, the quantizer 262generates the sale factor and the quantized spectral data by Formula 1.

$\begin{matrix}{{X_{ori} \cong X_{quant}} = {2^{\frac{scalefactor}{4}} \times {spectral\_ data}^{\frac{x + 1}{x}}}} & \left\lbrack {{Formula}\mspace{14mu} 1} \right\rbrack\end{matrix}$

In Formula 1, the ‘x’ can be a constant number that is an integerbetween 3 and 7.

The entropy coding unit 263 performs entropy coding on the scale factorand the quantized spectral data. The entropy coding unit 263 encodes thescale factor and the quantized spectral data by Huffman coding orarithmetic coding, and preferably, by the arithmetic coding, by whichthe present invention is non-limited.

Hence, if the input signal passes through the LPC analysis unit 240 andthen moves to a right path, configuration and function of the TCXencoding unit 260 become similar to those of the A encoding unit 130shown in FIG. 1.

Subsequently, the multiplexer 270 generates at least one or morebitstreams by multiplexing the linear prediction coding degreeinformation and linear prediction coding coefficients generated by theLPC analysis unit 240 with the data encoded by the ACELP encoding unit250 or the TCX encoding unit 260 and is then able to transmit thegenerated at least one or more bitstreams.

In the following description, a method for the LPC analysis unit togenerate linear prediction coding coefficients of variable degree in thesignal encoding apparatus according to the second embodiment of thepresent invention is explained in detail with reference to FIG. 3.

FIG. 3 is a detailed block diagram of an LPC analysis unit of a signalencoding apparatus according to a second embodiment of the presentinvention.

Referring to FIG. 3, the LPC analysis unit 240 includes an LPC degreedetermining unit 241, an LPC coefficients determining unit 242 and anLPC residual generating unit 243.

First of all, in case of receiving an input signal, the LPC degreedetermining unit 241 obtains property of the input signal and is thenable to determine a degree of a linear prediction coding coefficientthat will be generated by linear prediction coding. In this case, theproperty of the input signal can be determined in consideration of anextent of the tonal included in the corresponding signal and a spectraltilt degree. Preferably, the property of the input signal can bedetermined according to whether the input signal is suitable for thelinear prediction coding of high degree. If the input signal is a speechsignal having a lot of voiced sound or an audio signal having atonal-strong property, the LPC degree determining unit 241 generateslinear prediction coding degree information indicating a higher degree.On the contrary, if the input signal is a speech signal having a lot ofunvoiced sound or a noise-like audio signal, the LPC degree determiningunit 241 generates linear prediction coding degree informationindicating a lower degree.

Meanwhile, for another instance, if the input signal is a speech-likesignal, a linear prediction coding coefficient of high degree isgenerated. If the input signal is an audio-like signal, a linearprediction coding coefficient of low degree is generated. Hence, the LPCdegree determining unit 241 generates linear prediction codinginformation indicating a higher degree if the input signal is aspeech-like signal. On the contrary, the LPC degree determining unit 241generates linear prediction coding information indicating a lower degreeif the input signal is an audio-like signal.

The linear prediction coding degree information is able to indicate adegree of an integer for a linear prediction coding coefficient, andpreferably, a degree represented as Formula 2, and more preferably, adegree represented as Formula 3.

Degree (n) indicated by LPC_degree_information∈{N|0, 2, 4, . . . , 12,14, 16}  [Formula 2]

Degree (n) indicated by LPC_degree_information∈{N|0, 4, 8, . . . , 12,16, 32}  [Formula 3]

As mentioned in the above description, if an input signal is lesssuitable for LPC modeling, linear prediction coding degree informationindicating a lower degree is generated. Hence, it is able to select adegree of such a low number as 0, 2 or 0, 4. In case that the inputsignal include a noise-like audio signal only or a speech signalincluding an unvoiced sound only, the linear prediction coding degreeinformation can include the information indicating that a degree of thelinear prediction coding coefficient is 0.

Meanwhile, in case that the linear prediction coding degree informationis determined according to whether an input signal is an audio-likesignal or a speech-like signal, when the input signal includes an audiosignal only, the linear prediction coding degree information can includethe information indicating that a degree of the linear prediction codingcoefficient is 0.

In case of indicating a degree represented as Formula 2, the linearprediction coding degree information can be represented using 4 bits. Incase of indicating a degree represented as Formula 3, the linearprediction coding degree information can be represented using 3 bits.

As the linear prediction coding degree information indicates the numberof previous signals used to predict a signal of a current frame in thelinear prediction coding, if the linear prediction coding degreeinformation indicates a bigger integer, it can be observed that thenumber of the previous signals becomes higher. In particular, asmentioned in the foregoing description, if an input signal is a speechsignal having a lot of voiced sound or a tonal-strong audio signal, ahigh degree is indicated. In this case, it can be recognized that thenumber of the previous signals used for the linear prediction coding israised. On the contrary, if an input signal is a speech signal having alot of unvoiced sound or a noise-like audio signal, the number of theprevious signals used for the linear prediction coding will be lowered.

Afterwards, the input signal, of which linear prediction coding degreeis determined, is inputted to the LPC coefficients determining unit 242.Subsequently, based on the linear prediction coding degree informationdetermined by the LPC degree determining unit 241, the LPC coefficientsdetermining unit 242 is able to determine linear prediction codingcoefficients from the input signal. If the input signal is a speechsignal having a lot of unvoiced signal or a more noise-like audiosignal, a smaller number of LPC coefficients will be determined based onthe linear prediction coding degree information. If the input signal isan audio signal having a lot of voiced sound or a tonal-strong audiosignal, a more number of LPC coefficients will be determined based onthe linear prediction coding degree information.

Afterwards, the LPC residual generating unit 243 calculates a signal ofa difference between the input signal and the linear prediction codingsignal calculated by the LPC coefficients determining unit 242 in thecourse of determining the linear prediction coding coefficients and isthen able to output the calculated difference signal as an LPC residual.In this case, if the input signal is a speech signal having a lot ofunvoiced sound or a more noise-like audio signal, a smaller number oflinear prediction coding coefficients are used. Hence, the LPC codedsignal can be different from an original input signal and the LPCresidual will be similar to the original input signal. On the contrary,if the input signal is a speech signal having a lot of voiced sound or atonal-strong audio signal, a greater number of linear prediction codingcoefficients are used. Hence, it is highly probable that the LPCresidual can becomes a small signal different from the original inputsignal. In this case, if the LPC residual generated by performing thelinear prediction coding is the small signal, it can mean that the inputsignal is more suitable for the LPC modeling.

The second embodiment of the present invention pays attention to asignal coding method and apparatus in case that the linear predictioncoding degree information indicates a lower degree.

In this case, since the LPC residual outputted from the LPC residualgenerating unit 243 is a signal close to the input signal inputted tothe LPC analysis unit 240, if the input signal is an audio signal, thesignal encoded through the TCX encoding unit 260 shown in FIG. 2 has theproperty similar to that of the signal encoded through the MDCT encodingunit 130 shown in FIG. 1.

In particular, since the LPC analysis unit 240 according to the secondembodiment of the present invention determines the linear predictioncoding degree not as the fixed degree but as the variable degreeaccording to the signal property, it is able to efficiently code aspeech or audio signal, of which LPC modeling is difficult, vulnerableto distortion in case of applying the linear prediction coding of thefixed degree.

Moreover, since the TCX encoding unit 240 according to the secondembodiment of the present invention have the same configuration andfunction o the A encoding unit 130 (i.e., the MDCT encoding unit)according to the first embodiment of the present invention, the presentinvention provides an effect of coding an audio signal and a speechsignal effectively using two modules including the ACELP encoding unit250 and the TCX encoding unit 260 only.

FIG. 4 is a schematic block diagram of a signal decoding apparatus 200Acorresponding to the signal encoding apparatus 100A shown in FIG. 2according to the second embodiment of the present invention.

Referring to FIG. 4, a signal decoding apparatus 200A mainly includes ademultiplexer 410, an ACELP decoding unit 440, a TCX decoding unit 450,a signal compensating unit 460 and an LPC synthesis unit 470.

The demultiplexer 410 receives an input of the at least one or morebitstreams transmitted from the multiplexer 270 of the signal encodingapparatus 100A shown in FIG. 2 and then extracts an LPC bitstreamincluding the linear prediction coding degree information and the linearprediction coding coefficients, a first data encoded by ACELP scheme ora second data encoded by TCX scheme, and coding mode informationindicating the corresponding coding scheme. In this case, the first orsecond data can include the data outputted from the ACELP encoding unit250 described with reference to FIG. 2 or the data outputted from theTCX encoding unit 260 described with reference to FIG. 2.

The first/second data can be decoded by the ACELP/TCX decoding unit410/450 based on the coding mode information. The ACELP decoding unit410 performs a general ACELP decoding scheme. And, the TCX decoding unit450 includes an entropy decoding unit 451, a dequantizer 452 and aninverse MDCT unit 453.

The TCX decoding unit 450 according to the second embodiment of thepresent invention is the unit corresponding to the TCX encoding unit 260shown in FIG. 2. The entropy decoding unit 451 decodes the quantizeddata from the scale factor and spectral data included in the seconddata. Subsequently, the dequantizer 452 dequantizes the quantized data.The inverse MDCT unit 453 receives the dequantized data and is then ableto reconstruct the LPC residual. Although the TCX decoding unit 450 ofthe present invention reconstructs the LPC residual, a detailed unit ofthe TCX decoding unit 450 have the configuration and function similar tothose of the A decoding unit 230 of the decoder 200 described withreference to FIG. 1. Therefore, according to an LPC applied extent, theTCX decoding unit 450 is able to perform the same function of the Adecoding unit 230 or the C decoding unit 250 shown in FIG. 1.

Therefore, the signal decoding apparatus 200A according to the presentinvention is able to efficiently code an audio signal, a speech signaland an audio signal mixed with a speech signal using two modules. Thisshall be described in detail with reference to FIGS. 5 to 7 later inthis disclosure.

Meanwhile, the first/second data (hereinafter named a decoded signal)decoded through the ACELP/TCX decoding unit 440/450 is inputted to thesignal compensating unit 460.

The signal compensating unit 460 is a unit configured to perform timedomain aliasing cancellation (hereinafter abbreviated TDAC) to preventdistortion of a signal generated from a contiguous part of signalsrespectively decoded by different schemes. This is attributed to thefollowing reasons. First of all, the TCX coding scheme is a scheme ofapplying a non-rectangular window. Secondly, the ACELP coding scheme isthe scheme of applying a rectangular window. Thus, since each codingscheme uses a window of a different type, if signals respectivelydecoded by different coding schemes are contiguous, such defect asaliasing and the like can be generated due to asymmetry within therectangular window and the non-rectangular window are overlapped witheach other. The signal compensating unit 460 compensates for this defectusing folding, unfolding, windowing, compensation information and thelike. This shall be described in detail with reference to FIGS. 8 to 10later in this disclosure.

Afterwards, the LPC synthesis unit 470 is able to reconstruct anoriginal signal by receiving the decoded signal and the LPC bitstreamfrom the signal compensating unit 460 and the demultiplexer 410,respectively. The LPC synthesis unit 470 can vary an extent of applyingto the decoded signal according to the linear prediction coding degreeinformation and the linear prediction coding coefficients included inthe LPC bitstream. Therefore, the TCX decoding unit 450 is able toefficiently reconstruct an audio signal, of which main component is theaudio property difficult for the second decoding unit 220-2 includingthe general TCX decoding unit 250 to efficiently decode, as well as anaudio signal included in a speech signal. The detailed function and roleof the LPC synthesis unit 470 are explained with reference to FIG. 5 andFIG. 6 as follows.

FIG. 5 is a detailed block diagram of the LPC synthesis unit 470 shownin FIG. 4.

Referring to FIG. 5, the LPC synthesis unit 470 includes a linearprediction coding degree information decoding unit 471, a linearprediction coding coefficients extracting unit 472 and a signalsynthesis unit 473. First of all, the linear prediction coding degreeinformation decoding unit 471 receives an input of an LPC bitstream fromthe multiplexer 410 and then extracts the linear prediction codingdegree information from the LPC bitstream. In this case, the linearprediction coding degree information is identical to the contentsdescribed with reference to FIG. 3. If a decoded signal is a speechsignal having a lot of unvoiced sound or a more noise-like audio signal,the linear prediction coding degree information will indicate a lowerdegree.

The linear prediction coding coefficients extracting unit 472 extractslinear prediction coding coefficients from the LPC bitstream based onthe linear prediction coding degree. In this case, since the linearprediction coding degree is proportional to the number of the linearprediction coding coefficients, if the decoded signal is the speechsignal having a lot of unvoiced sound or the more noise-like audiosignal, it is a matter of course that the less number of the linearprediction coding coefficients are extracted.

The signal synthesis unit 473 generates an output signal byreconstructing an original signal by applying the linear predictioncoding coefficients to the decoded signal.

Thus, the signal decoding apparatus 200A according to the secondembodiment o the present invention is able to perform the functions ofthe three modules (i.e., the A decoding unit 230, the B decoding unit240 and the C decoding unit 250) of the decoder 200 shown in FIG. 1using the two modules (i.e., the ACELP decoding unit 440 and the TCXdecoding unit 450) in a manner that the LPC synthesis unit 470 appliesthe variable linear prediction coding coefficients using the linearprediction coding degree information variably determined according tothe signal property.

Therefore, since the window switchings less than the window switchingsamong three modules are generated, it is also able to reduce the signaldistortions caused by the window switchings. Moreover, in case of codinga signal in which a speech signal and an audio signal are mixedtogether, the signal can be coded using a less number of modules.Therefore, it is advantageous in simplifying the corresponding signalprocessing apparatus.

Moreover, when the LCP synthesis unit 270 shown in FIG. 1 applies thelinear prediction coding of a fixed degree, it is able to efficientlycode a speech or audio signal vulnerable to distortion, i.e., notsuitable for the LPC modeling.

FIG. 6 is a flowchart of a signal decoded in accordance with linearprediction degree information in the LPC synthesis unit 470 shown inFIG. 5. As functions of a linear prediction coding degree informationdecoding unit 671, a linear prediction coding coefficients extractingunit 672 and a signal synthesis unit 673 are equal to those of theformer units having the same names in FIG. 5, their details are omittedfrom the following description. Yet, note the examples shown below theunits, respectively.

First of all, in case that the linear prediction coding degreeinformation indicates that a linear prediction coding degree is 16degree [case of (a)], linear prediction coding coefficients extracted bythe linear prediction coding coefficients extracting unit 672 correspondto A={a_(1, a) _(2, a) ₃, . . . a₁₄, a₁₅, a₁₆}. Subsequently, the signalsynthesis unit 673 generates an output signal (Y=A•X) by applying thelinear prediction coding coefficients to the decoded signal (X). Sincethe LPC synthesis unit 270 of the second decoding unit 220-2 accordingto the first embodiment of the present invention for coding speech andaudio signals uses the linear prediction coding coefficients of the 16degree, when the LPC synthesis unit 470 uses the linear predictioncoding coefficients of the 16 degree [case of (a)], the signal decodingapparatus 200A according to the second embodiment of the presentinvention shown in FIG. 4 is able to perform the same function of thesecond decoding unit 220-2 of the first embodiment. In particular, incase of (a), an output signal can be generated in the same manner of themethod of reconstructing the output signal through the B decoding unit240 and the LPC synthesis unit 270 according to the first embodiment ofthe present invention (if the decoding signal (X) is a speech signal) orthe method of reconstructing the output signal through the C decodingunit 250 and the LPC synthesis unit 270 according to the firstembodiment of the present invention (if the decoded signal (X) is anaudio signal mixed in a speech signal).

Secondly, in case that the linear prediction coding degree informationindicates that a linear prediction coding degree is 4 degree [case of(b)], linear prediction coding coefficients extracted by the linearprediction coding coefficients extracting unit 672 correspond to B={b₁,b₂, b₃, b₄}. Subsequently, the signal synthesis unit 673 generates anoutput signal (Y=B•X) by applying the linear prediction codingcoefficients to the decoded signal (X).

Thirdly, in case that the linear prediction coding degree informationindicates that a linear prediction coding degree is 0 degree [case of(c)], linear prediction coding coefficients extracted by the linearprediction coding coefficients extracting unit 672 do not exist at allto result in C={). Subsequently, the signal synthesis unit 673 finallygenerates an output signal (Y=X) equal to the decoded signal by applyingthe linear prediction coding coefficients to the decoded signal (X). Inhe case of (c), since the linear prediction coding coefficients are notdecoded, it can be observed that the decoded signal (X) inputted to theLPC synthesis unit 470 bypasses the LPC synthesis unit 470.

The A decoding unit 230 according to the first embodiment of the presentinvention does not perform the linear prediction coding but decodes asignal through entropy decoding, dequantization and inverse MDCT.Therefore, when an input signal is decoded into an audio signal in thesignal decoding apparatus 200A according to the second embodiment of thepresent invention using the TCX decoding unit 450, if the LPC synthesisunit 470 uses the linear prediction coding coefficient of 0 degree,i.e., if none of the linear prediction coding coefficients is decoded[case of (c)], the right path in the signal decoding apparatus 200Aaccording to the second embodiment of the present invention shown inFIG. 4 can perform the same function of the first decoding unit 220-1 ofthe first embodiment.

Therefore, the signal decoding apparatus 200A according to the secondembodiment of the present invention uses two modules in a manner ofusing linear prediction coding coefficients of variable degree in theLPC coding, thereby performing the same function for the decoder 200according to the first embodiment of the present invention to code asignal generated from mixing audio and speech signals together usingthree modules.

FIG. 7 is a schematic block diagram of a path of a signal generatedthrough the signal decoding apparatus 200A if linear prediction degreeinformation of the present invention indicates 0 [case of (c)]. Asconfigurations and functions of an ACELP decoding unit 710, a TCXdecoding unit 720, a signal compensating unit 730 and an LPC synthesisunit 740 shown in FIG. 7 are equal to those of the former units havingthe same names in FIG. 4, their details are omitted from the followingdescription.

Referring to FIG. 7, if the linear prediction coding degree informationof the present invention indicates 0, as mentioned in the foregoingdescription with reference to FIG. 6, the linear prediction codingcoefficients are not decoded but the decoded signal (X) outputted fromthe signal compensating unit 730 becomes an output signal as it is.Hence, the LPC synthesis is not performed (i.e., the LPC synthesis unitis non-activated). In this case, as mentioned in the foregoingdescription, the right path in the signal decoding apparatus 200Aaccording to the second embodiment of the present invention shown inFIG. 7 can perform the same function of the first decoding unit 220-1 ofthe first embodiment.

Meanwhile, as the second decoding unit 220-2 according to the firstembodiment of the present invention uses the linear prediction codingcoefficients of fixed degree, LPC modeling is essentially performed on asignal not suitable for the LPC modeling. Therefore, it is highlyprobable that signal distortion may occur. Yet, the signal decodingapparatus 200A according to the second embodiment of the presentinvention transmits linear prediction coding degree information of lowdegree for a signal not suitable for the LPC modeling. In particular,the signal decoding apparatus 200A according to the second embodiment ofthe present invention extremely transmits the linear prediction codingdegree information set to 0. Therefore, the signal decoding apparatus200A according to the second embodiment of the present invention is ableto considerably reduce the signal distortion because the speech or audiosignal is decoded using the ACELP decoding unit 440 or the TCX decodingunit 450 only.

In order to prevent the signal distortion generated from a contiguouspart of signals respectively decoded by different schemes, the signalcompensating unit 730 of the signal decoding apparatus 200A according tothe second embodiment of the present invention performs time domainaliasing cancellation (TDAC). This is explained with reference to FIGS.8 to 10 as follows.

FIG. 8 is a diagram for various examples of applying a different codingscheme per frame (or subframe) to an input signal configured by a blockunit according to the present invention.

Referring to FIG. 8, an input signal is configured with a series offrames including an n^(th) frame (Frame n), an (n+1)^(th) frame (Framen+1) and the like. Moreover, one frame (e.g., n^(th) frame, (n+1)^(th)frame, etc.) is configured with a plurality of subframes (e.g., 4subframes in FIG. 8). Different coding schemes, as shown in FIG. 8, canbe applied to the subframes, respectively. In this case, the codingschemes can include ACELP coding schemes and TCX coding schemes.Referring to FIG. 8( a) to FIG. 8( c), it can be observed that the ACELPcoding scheme is applied by one subframe unit. Meanwhile, the TCX codingscheme is applied by one subframe unit [(a)], is applied to twocontiguous subframes [(b)], and can be applied to four contiguoussubframes [(c)], by which the present invention is non-limited. If theTCX coding scheme is applied to the four contiguous subframes, it meansthat the TCX coding scheme can be applied to one frame. Thus, it can beobserved that a length of a block of an audio signal using the TCXcoding scheme of the present invention is preferably set to an integermultiple of a length of a block of a speech signal for applying theACELP coding scheme, and preferably, to a multiple of (2*integer).

FIG. 9 is a diagram of blocks, to which different coding schemes areapplied, among the contiguous frames (or subframes) shown in FIG. 8.

Referring to FIG. 9, since each coding scheme uses a different window,the ACELP coding scheme can be classified into a rectangular codingscheme according to a type of window and the like. And, the TCX codingscheme can be classified into a non-rectangular coding scheme. Asmentioned in the foregoing description with reference to FIG. 8, the TCXcoding scheme is applicable to one subframe, two contiguous subframes orfour contiguous subframes (i.e., one frame). In this case, a unit ofapplying the non-rectangular coding scheme or the rectangular codingscheme shall be named a block. In other words, the block can correspondto the one frame, the one subframe or the two contiguous subframes.

The blocks, to which the different coding schemes are applied,respectively, can be mainly categorized into two kinds of cases. Firstof all, the ACELP coding scheme (i.e., the rectangular coding scheme) isswitched to the TCX coding scheme (i.e., the non-rectangular codingscheme). Secondly, the TCX coding scheme (i.e., the non-rectangularcoding scheme) is switched to the ACELP coding scheme (i.e., therectangular coding scheme). Referring to FIG. 9, in the first case ofswitching the ACELP coding scheme to the TCX coding scheme, thecorresponding contiguous blocks are indicated by dotted lines. Forinstance, referring to FIG. 9( a), the corresponding contiguous blocksinclude a third block and a fourth block.

On the contrary, in the case of switching the TCX coding scheme to theACELP coding scheme, the indication is omitted from FIG. 9. Yet, thecorresponding contiguous blocks such as 4^(th) and 5^(th) blocks existeverywhere.

Thus, since the different coding schemes are used, such defect asaliasing can be generated from the part, at which the rectangular windowand the non-rectangular window are overlapped with each other, due toasymmetry. And, the defect generating process and a defect compensatingmethod are explained with reference to FIG. 10 as follows.

FIG. 10 is a diagram for a processing method when different type windowsare overlapped with each other in the frames, to which the differentcoding schemes are applied, respectively.

Referring to FIG. 10, it can be observed that a rectangular window and anon-rectangular window are overlapped with each other in a block C.Regarding a signal configured with bocks A to F, the rectangular windowis applied to the block B and the block C. And, the non-rectangularwindow is applied to the blocks C to F. FIG. 10( a) to FIG. 10( d) showthe results from sequentially applying windowing, folding, unfolding andwindowing to the blocks A to F, respectively. The windowing, folding,unfolding and windowing indicate the processes sequentially applied tothe blocks, respectively, in order to apply the time domain aliasingcancellation (TDAC) in association with the non-rectangular window.

Referring to FIG. 10( a), the results (i.e., dotted blocks) fromapplying the rectangular window to the block B and the block C arerepresented on the top. And, the results from applying thenon-rectangular window are represented on the bottom. In this case,C(L₁) indicates a result from applying a part L₁ of the non-rectangularwindow to the block C. And, D(L₂) indicates a result from applying apart L₂ of the non-rectangular window to the block D. Subsequently, ifthe folding is performed on the results of the non-rectangular windowapplied blocks C to F, it is able to obtain the results shown in FIG.10( b). In this case, Er or Dr indicates ‘reverse’ as the folding isperformed with reference to a block boundary. Afterwards, a result fromperforming the unfolding is shown in FIG. 10( c). Finally, if thenon-rectangular window is applied to the unfolded block, it is able toobtain the result shown in FIG. 10( d).

An uncompensated signal corresponding to an original signal of the blockD, i.e., a signal obtained from a transmitted data only can berepresented by Formula 4, as shown in FIG. 10( d).

uncompensated_signal=(−Cr(L ₁)r+D(L ₂))(L ₂)  [Formula 4]

In Formula 4, ‘C’ indicates a data corresponding to the block C, ‘D’indicates a data corresponding to the block D, ‘r’ indicates areversion, ‘L₁’ indicates a result from applying a part L₁ of thenon-rectangular window, and ‘L₂’ indicates a result from applying a partL₂ of the non-rectangular window.

As mentioned in the foregoing description, in order to prevent signaldistortion, the obtained uncompensated signal requires compensation.Therefore, a compensating signal for compensating the uncompensatedsignal to become identical or similar to an original signal iscalculated and transmitted. And, the compensating signal will be used bythe signal compensating unit 460 of the signal decoding apparatus 200A.In doing so, a method of calculating the compensating signal generallyfollows a method of processing a wide-sense audio signal having speechand audio signal mixed therein and its details shall be omitted from thefollowing description.

In the following description, applications including the signal encodingapparatus 100A or the signal decoding apparatus 200A according to thesecond embodiment of the present invention are described with referenceto FIGS. 11 to 14.

FIG. 11 is a block diagram of an encoder 1100 including the signalencoding apparatus 100A according to the second embodiment of thepresent invention. And, FIG. 12 is a block diagram of a decoder 1200including the signal decoding apparatus 200A according to the secondembodiment of the present invention.

Referring to FIG. 11, an encoder 1100 includes a plural channel encodingunit 1110, a band extension encoding unit 1120, a signal encodingapparatus 100A according to a second embodiment of the presentinvention, and a multiplexer 1160.

First of all, a downmix signal generated from downmixing an inputtedplural channel signal by the plural channel encoding unit 110 is named afull-range downmix signal. And, after a high frequency band signal isremoved from the full-range down mix signal, a downmix signal, in whicha low frequency band exists, is named a low frequency band downmixsignal.

The plural channel encoding unit 110 receives an input of a pluralchannel signal. In this case, the plural channel signal means a signalhaving at least three channels in general and is able to include a monosignal or a stereo signal. The plural channel encoding unit 1110generates a full-range downmix signal by downmixing the inputted pluralchannel signal and also generates spatial information necessary to upmixthe full-range downmix signal into a plural channel signal. In thiscase, the spatial information can include at least one of channel leveldifference information, channel prediction coefficients, inter-channelcorrelation information, downmix gain information and the like. If theplural channel encoding unit 1110 receives an input of a mono signal,downmixing is not performed and the mono signal can bypass the pluralchannel encoding unit 110.

The band extension encoding unit 1120 receives the full-range downmixsignal and is then able to generate spectral data corresponding to a lowfrequency band in the full-range downmix signal and extensioninformation corresponding to a signal of a frequency band. The extensioninformation is the information for the decoder stage to reconstruct thelow frequency band downmix signal, from which the frequency band isremoved, into the full-range downmix signal. And, the extensioninformation can be transmitted together with the spatial information.

The input signal is determined to be coded by a specific scheme based onsignal property. And, coding mode information indicating the codingscheme is generated [not shown in the drawing]. In this case, asmentioned in the foregoing descriptions with reference to FIG. 2 andFIG. 3, the coding scheme can include the ACELP coding scheme or the TCXcoding scheme. As mentioned in the foregoing descriptions with referenceto FIG. 2 and FIG. 3, before the low frequency band downmix signal iscoded by the coding scheme, the linear prediction coding (LPC) isperformed using the variable degree.

First of all, the linear prediction coding degree is determinedaccording to property of the low frequency band downmix signal inputtedto the LPC analysis unit 1130. Based on the determined linear predictioncoding degree, linear prediction coding coefficients and an LPC residualare then generated by performing the linear prediction coding.

Subsequently, the LPC residual is encoded by the coding schemedetermined according to the coding mode information.

If a specific frame or segment of the low frequency band downmix signalhas a dominant speech property, the ACELP encoding unit 1140 performsencoding by the ACELP scheme. In this case, the ACELP scheme may followthe AMR-WB (adaptive multi-rate wide-band) standard, by which thepresent invention is non-limited. Since the signal inputted to the ACELPencoding unit 1140 can have high redundancy on a time axis, modeling ispossible by the linear prediction that predicts a current signal from apast signal. Therefore, if the linear prediction coding scheme isadopted, coding efficiency can be raised. Moreover, the ACELP encodingunit 1140 can correspond to a time domain encoder.

In case of attempting to code a signal, in which a specific frame orsegment of a low frequency band downmix signal has an audio-dominantproperty, or in case of attempting to code an audio signal in the signalhaving audio and speech properties are mixed, the TCX encoding unit 1150is selected to encode the corresponding signal. In doing so, the TCXcoding scheme can include the scheme of performing frequency transformon the LPC residual obtained from performing the linear predictioncoding. In this case, the frequency transform can be performed by MDCT(modified discrete cosine transform), AAC (advanced audio coding)standard or HE-AAC (high efficiency advanced audio coding) standard, bywhich the present invention is non-limited. In particular, as mentionedin the foregoing descriptions with reference to FIG. 2 and FIG. 3, sincethe LPC analysis unit 1130 performs the linear prediction coding of thevariable degree, the TCX encoding unit 1150 of the present invention isable to play a role as a signal coding unit according to the AACstandard or the HE-AAC standard. For instance, if a specific frame orsegment of a low frequency band downmix signal has a dominant audioproperty, the LPC analysis unit 1130 performs the linear predictioncoding of low degree, or may not perform the linear prediction codingextremely. Instead, the corresponding encoding is performed by the TCXencoding unit 1150. Hence, the same function of the signal coding unitaccording to the AAC standard or the HE-AAC standard can be performed.

The multiplexer 1160 generates at least one or more bitstreams bymultiplexing the spatial information, the band extension information,the data encoded by the ACELP encoding unit 1140, the data encoded bythe TCX encoding unit 1150 and the like together and then transmits thegenerated at least one or more bitstreams.

Referring to FIG. 12, the decoder 1200 includes a demultiplexer 1210, asignal decoding apparatus 220A, a band extension decoding unit 1260 anda plural channel decoding unit 1270.

The demultiplexer 1210 extracts the encoded signal data, band extensioninformation, spatial information and the like encoded from the bitstreamtransmitted by the encoder.

The signal decoding apparatus 200A including an ACELP decoding unit1220, a TCX decoding unit 1230, a signal compensating unit 1240 and anLPC synthesis unit 1250 has the same configurations and functionsdescribed with reference to FIGS. 4 to 10 and its details are omittedfrom the following description.

By performing a band extension decoding scheme on an output signal fromthe signal decoding apparatus 200A using the band extension information,the band extension decoding unit 1260 reconstructs a downmix signal of ahigh frequency band and is able to output a full-range downmix signal.

In doing so, it is able to generate the full-range downmix signal usingthe whole low frequency band downmix signal and the band extensioninformation or using the low frequency band downmix signal in part.

The plural channel decoding unit 1270 is able to generate aplural-channel output signal (e.g., stereo signal included) by applyingthe spatial information to the full-range downmix signal.

The signal processing apparatus according to the second embodiment ofthe present invention is available for various products to use. Thesesproducts can be mainly grouped into a stand alone group and a portablegroup. A TV, a monitor, a settop box and the like can be included in thestand alone group. And, a PMP, a mobile phone, a navigation system andthe like can be included in the portable group.

FIG. 13 shows relations between products, in which a signal processingapparatus 200A according to a second embodiment of the present inventionis implemented.

Referring to FIG. 13, a wire/wireless communication unit 1310 receives abitstream via wire/wireless communication system. In particular, thewire/wireless communication unit 1310 can include at least one of a wirecommunication unit 1310A, an infrared unit 1310B, a Bluetooth unit 1310Cand a wireless LAN unit 1310D.

A user authenticating unit 1320 receives an input of user informationand then performs user authentication. The user authenticating unit 1320can include at least one of a fingerprint recognizing unit 1320A, aniris recognizing unit 1320B, a face recognizing unit 1320C and a voicerecognizing unit 1320D. The fingerprint recognizing unit 1320A, the irisrecognizing unit 1320B, the face recognizing unit 1320C and the speechrecognizing unit 1320D receive fingerprint information, irisinformation, face contour information and voice information and thenconvert them into user informations, respectively. Whether each of theuser informations matches pre-registered user data is determined toperform the user authentication.

An input unit 1330 is an input device enabling a user to input variouskinds of commands and can include at least one of a keypad unit 1330A, atouchpad unit 1330B and a remote controller unit 1330C, by which thepresent invention is non-limited.

A signal coding unit 1340 performs encoding or decoding on an audiosignal and/or a video signal, which is received via the wire/wirelesscommunication unit 1310, and then outputs an audio signal in timedomain. The signal coding unit 1340 includes an audio signal processingapparatus 1345. As mentioned in the foregoing description, the audiosignal processing apparatus 1345 corresponds to the signal encodingapparatus 100A or the signal decoding apparatus 200A according to thesecond embodiment of the present invention. Thus, they can beimplemented by at least one or more processors.

A control unit 1350 receives input signals from the input unit 1330 andcontrols all processes of the signal decoding unit 1340 and an outputunit 1360. In particular, the output unit 1360 is a component configuredto output an output signal generated by the signal decoding unit 1340and the like and can include a signal output unit 1360A and a displayunit 1360B. If the output signal is an audio signal, it is outputted viathe signal output unit 1360A. If the output signal is a video signal, itis outputted via the display unit 1360B.

FIG. 14 is a diagram for explaining relations between products in whicha signal processing apparatus according to the present invention isimplemented, in which the relation between a terminal and server for theproducts shown in FIG. 13 is shown.

Referring to FIG. 14(A), it can be observed that a first terminal 1410and a second terminal 1420 can exchange data or bitstreamsbi-directionally with each other via the wire/wireless communicationunits. Referring to FIG. 14( b), it can be observed that a server 1430and a first terminal 1440 can perform wire/wireless communications witheach other.

Thus, as the signal processing apparatus is included in a real product,linear prediction coding degree information indicating a variable degreeis used according to property of a signal. Therefore, using modules lessthan those of a general signal processor, it is able to efficiently codea signal having speech and audio signals mixed therein.

A decoding/encoding method according to the present invention can beimplemented into a computer-executable program and can be stored in acomputer-readable recording medium. And, multimedia data having a datastructure of the present invention can be stored in thecomputer-readable recording medium. The computer-readable media includeall kinds of recording devices in which data readable by a computersystem are stored. The computer-readable media include ROM, RAM, CD-ROM,magnetic tapes, floppy discs, optical data storage devices, and the likefor example and also include carrier-wave type implementations (e.g.,transmission via Internet). And, a bitstream generated by the abovementioned encoding method can be stored in the computer-readablerecording medium or can be transmitted via wire/wireless communicationnetwork.

INDUSTRIAL APPLICABILITY

Accordingly, the present invention is applicable to encoding anddecoding an audio signal.

While the present invention has been described and illustrated hereinwith reference to the preferred embodiments thereof, it will be apparentto those skilled in the art that various modifications and variationscan be made therein without departing from the spirit and scope of theinvention. Thus, it is intended that the present invention covers themodifications and variations of this invention that come within thescope of the appended claims and their equivalents.

1. A method of processing a signal, comprising: receiving coding modeinformation indicating a speech coding scheme or an audio coding scheme,linear prediction coding degree information indicating a linearprediction coding degree, and the signal including at least one of aspeech signal and an audio signal,; decoding the signal according to thespeech coding scheme or the audio coding scheme based on the coding modeinformation; decoding linear prediction coding coefficients of thesignal based on the linear prediction coding degree information; andgenerating an output signal by applying the decoded linear predictioncoding coefficients to the decoded signal, wherein the linear predictioncoding degree information is determined based on a variation of a valueof an LPC residual generated from performing the linear predictioncoding on the signal.
 2. The method of claim 1, wherein if the signal isthe speech signal having a lot of voiced sound, the linear predictioncoding degree information indicates a degree higher than that of thespeech signal having a lot of unvoiced sound.
 3. The method of claim 1,wherein if the signal is the audio signal having a strong tonalcomponent, the linear prediction coding degree information indicates adegree higher than that of the audio signal having a week tonalcomponent.
 4. The method of claim 1, wherein if the signal is anaudio-like signal, the linear prediction coding degree informationindicates a degree lower than that of a speech-like signal.
 5. Themethod of claim 1, wherein a frame length of the audio signal is aninteger multiple of a frame length of the speech signal.
 6. The methodof claim 1, further comprising: if the decoded signal is the speech oraudio signal and a signal of a previous frame is different from thedecoded signal, preventing aliasing by compensating the decoded signalusing the signal of the previous frame.
 7. An apparatus for processing asignal, comprising: a multiplexer receiving an LPC bitstream includinglinear prediction coding degree information indicating a linearprediction coding degree and linear prediction coding coefficients, thesignal including at least one of a speech signal and an audio signal,and coding mode information indicating a speech coding scheme or anaudio coding scheme; an ACELP decoding unit decoding the signalaccording to the speech coding scheme if the coding mode informationindicates the speech coding scheme; a TCX decoding unit decoding thesignal according to the audio coding scheme if the coding modeinformation indicates the audio coding scheme; and an LPC synthesis unitdecoding the linear prediction coding coefficients of the signal basedon the linear prediction coding degree information, and generating anoutput signal by applying the decoded linear prediction codingcoefficients to the decoded signal, wherein the linear prediction codingdegree information is determined based on a variation of a value of anLPC residual generated from performing the linear prediction coding onthe signal.
 8. The apparatus of claim 7, wherein if the signal is thespeech signal having a lot of voiced sound, the linear prediction codingdegree information indicates a degree higher than that of the speechsignal having a lot of unvoiced sound.
 9. The apparatus of claim 7,wherein if the signal is the audio signal having a strong tonalcomponent, the linear prediction coding degree information indicates adegree higher than that of the audio signal having a week tonalcomponent.
 10. The apparatus of claim 7, wherein the LPC synthesis unitcomprises: an LPC degree information decoding unit decoding the linearprediction coding degree information from the LPC bitstream; an LPCcoefficients extracting unit extracting the linear prediction codingcoefficients from the LPC bitstream based on the linear predictioncoding degree information; and a signal synthesis unit generating theoutput signal by applying the linear prediction coding coefficients tothe decoded signal.
 11. The apparatus of claim 10, if the linearprediction coding degree information indicates that the linearprediction coding coefficients are not decoded, wherein the decodedsignal bypasses the signal synthesis unit and is then outputted.
 12. Theapparatus of claim 7, further comprising: if a coding scheme of thedecoded signal adopts the coding scheme different from that of thedecoded signal of a previous frame, a signal compensating unitpreventing aliasing by compensating the decoded signal using the decodedsignal of the previous frame.
 13. The apparatus of claim 7, wherein aframe length of the audio signal is an integer multiple of a framelength of the speech signal.
 14. A method of processing a signal,comprising: determining linear prediction coding degree informationindicating a variable degree of linear prediction coding coefficientsaccording to property of an input signal; determining the linearprediction coding coefficients based on the linear prediction codingdegree information; generating an LPC residual using the linearprediction coding coefficients and the input signal; and encoding theLPC residual using either an audio coding scheme or a speech codingscheme, wherein the linear prediction coding degree information isdetermined based on a variation of a value of an LPC residual generatedfrom performing the linear prediction coding on the input signal.
 15. Anapparatus for processing a signal, comprising: an LPC analysis unitdetermining linear prediction coding degree information indicating avariable degree of linear prediction coding coefficients according toproperty of an input signal, the LPC analysis unit determining thelinear prediction coding coefficients based on the linear predictioncoding degree information, the LPC analysis unit generating an LPCresidual using the linear prediction coding coefficients and the inputsignal; an ACELP encoding unit encoding the LPC residual using an audiocoding scheme; a TCX encoding unit encoding the LPC residual using aspeech coding scheme; and a multiplexer generating a bitstream includingthe linear prediction coding degree information, the linear predictioncoding coefficients and the coded signal, wherein the linear predictioncoding degree information is determined based on a variation of a valueof an LPC residual generated from performing the linear predictioncoding on the input signal.